June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] powerpc: select PPC_CLOCK unconditionally for FSL_SOC
by Shawn Guo 10 Jul '12

10 Jul '12

Freescale PowerPC SoCs share a number of IP blocks with Freescale ARM/IMX SoCs, FlexCAN, SSI, FEC, eSDHC, USB, etc. There are some effort consolidating those drivers to make them work for both architectures. One outstanding difference between two architectures is ARM/IMX will turn off module clocks during platform initialization for power saving and expects drivers manage clocks using clk API, while PowerPC mostly does not do that, and thus does not always build in clk API. Listing all those driver Kconfig options in "select PPC_CLOCK if" seems not scalable for long term maintenance, and could easily introduce Kconfig recursive dependency. This patch chooses to select PPC_CLOCK unconditionally for FSL_SOC to always build clk API for PowerPC in. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- arch/powerpc/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/Kconfig b/arch/powerpc/Kconfig index feab3ba..63fa7fb 100644 --- a/arch/powerpc/Kconfig +++ b/arch/powerpc/Kconfig @@ -661,7 +661,7 @@ config SBUS config FSL_SOC bool select HAVE_CAN_FLEXCAN if NET && CAN - select PPC_CLOCK if CAN_FLEXCAN + select PPC_CLOCK config FSL_PCI bool -- 1.7.5.4

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[alsa-devel] A problem with USB audio devices and the UHCI scheduler
by Monty Montgomery 04 Jul '12

04 Jul '12

A question for Clemens et al... Background: I can run a USB audio device at 6ch/24 bit single duplex, but I cannot run the same device at 16 bit stereo full duplex. UHCI and EHCI both say 'not enough bandwidth' (OHCI works). EHCI's scheduling issues are well known, but I was surprised UHCI could not schedule this either. The cause is simple; UHCI (and EHCI) use the endpoint's wMaxPacketSize for all bandwidth allocation. My device supports all of its sample rates on the same altset, with a wMaxPacketSize matching the highest (96kHz) rate. So, the HCD is reserving most of the USB bus's bandwidth regardless of the actual requirements. How should I approach fixing this in the driver? There's currently no means to communicate a desired bandwidth reservation to the HCD from USB audio, so I'd have to extend even the quirks. I was considering shadowing the endpoint's original wMaxPacketSize in the substream struct, and replacing the value in the endpoint's native descriptor while the stream is running. I have no idea how much trouble that would cause (eg, corruption elsewhere, or middle finger from Linus). Suggestions? For now I have things working by disabling UHCI's initial bandwidth checks (as there is in fact more than enough bandwidth to schedule what I'm doing) but I'd like a more permanent solution. Monty

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add deemphasis switch
by Prchal Jiří 04 Jul '12

04 Jul '12

This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4ed6c20..2265e94 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -381,6 +381,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), + /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0, adc_tlv), -- 1.7.7

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[alsa-devel] [PATCH V3 0/4] Add support for STA529 codec, spdif_receiver and SPEAr spdif_out controller
by Rajeev Kumar 03 Jul '12

03 Jul '12

Earlier I had circulated Patch set V2. Please find below the link for the same. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052868.html. Some of the patches got applied and for rest I am sending Version V3. Patches that got applied(from V2 of patch set) are listed below: 1. [PATCH V2 3/9] sound:asoc: Add support for synopsys i2s controller as per ASoC framework. 2. [PATCH V2 4/9] sound:asoc: Add support for SPEAr ASoC pcm layer. 3. [PATCH V2 5/9] sound:asoc:spdif_in: Add spdif IN support Changes since V2: 1. In-corporated review comments received on V2 Rajeev Kumar (2): ASoC: STA529: Add support for STA529 Audio Codec ASoC: SPEAr: Add Kconfig and Makefile to support SPEAr audio driver Vipin Kumar (2): ASoC: spdif_receiver: Add support for spdif in Audio Codec ASoC: SPEAr spdif_out: Add spdif out support sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 67 ++++++ sound/soc/codecs/sta529.c | 441 +++++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 22 ++ sound/soc/spear/Makefile | 4 + sound/soc/spear/spdif_out.c | 387 ++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 +++++++ 8 files changed, 1007 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h

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[alsa-devel] [PATCH] ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect
by Benoît Thébaudeau 03 Jul '12

03 Jul '12

snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with invert. In that case, the raw disconnect value should be max, which corresponds to the userspace value 0. This use case currently does not appear upstream, but it could break SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future. Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- .../sound/soc/soc-dapm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c index 405841c..5ef082f 100644 --- linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c +++ linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c @@ -2515,19 +2515,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int wi; val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val);

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[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 03 Jul '12

03 Jul '12

Second version of patch keeping formatting. Adds missing register default values to cache. Adds register and bits definitions in header file. Changes are for TLV320AIC310x based on data sheet. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -122,7 +122,9 @@ 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

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[alsa-devel] 2.6.18 unknown symbol ksize error
by Patrick Shirkey 02 Jul '12

02 Jul '12

Hi, I have compiled the latest stable alsa drivers against 2.6.18 kernel with gcc-4.1. I am getting an error when I insert the sound card modules. snd_rawmidi: Unknown symbol ksize As far as I can tell 2.6.18 has support for ksize so I am not sure where this error is coming from. -- Patrick Shirkey Boost Hardware Ltd

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[alsa-devel] alsa-driver cross compile problem - upgrading ALSA
by Max Klein 02 Jul '12

02 Jul '12

I am running 2.6.27 kernel on arm board with ALSA driver version 1.0.17 (cat /proc/asound/version) and I would like to update ALSA driver version to current 1.0.25. I am trying to build alsa-drivers with "./configure --with-cross=arm-none-linux-gnueabi- --enable-dynamic-minors --with-debug=full --with-isapnp=no --with-oss=no --with-kernel=/data/src/kernel/linux-davinci --with-build=/data/src/kernel/linux-davinci --with-sequencer=no" Afterwards I compile it via "make ARCH=arm" but this fails we the following error: In file included from /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/acore/hrtimer.c:1: /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:753: error: static declaration of 'jiffies_to_msecs' follows non-static declaration include/linux/jiffies.h:286: error: previous declaration of 'jiffies_to_msecs' was here /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:772: error: static declaration of 'msecs_to_jiffies' follows non-static declaration include/linux/jiffies.h:288: error: previous declaration of 'msecs_to_jiffies' was here Can someone please help me how to get ALSA upgraded on my platform? Obviously I am either doing something wrong when configuring or I am on the absolutely wrong way? I am looking forward for any advice, thanks in advance! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] [PATCH] mioa701 convert to snd_soc_register_card API
by Robert Jarzmik 01 Jul '12

01 Jul '12

The mioa701 board code is converted to the snd_soc_register_card() and snd_soc_unregister_card() APIs. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/pxa/mioa701_wm9713.c | 33 ++++++++++++--------------------- 1 file changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 9c585af..8687c1c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = { .num_links = ARRAY_SIZE(mioa701_dai), }; -static struct platform_device *mioa701_snd_device; - -static int mioa701_wm9713_probe(struct platform_device *pdev) +static int __devinit mioa701_wm9713_probe(struct platform_device *pdev) { - int ret; + int rc; if (!machine_is_mioa701()) return -ENODEV; - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" - "lead to overheating and possible destruction of your device." - "Do not use without a good knowledge of mio's board design!\n"); - - mioa701_snd_device = platform_device_alloc("soc-audio", -1); - if (!mioa701_snd_device) - return -ENOMEM; - - platform_set_drvdata(mioa701_snd_device, &mioa701); - - ret = platform_device_add(mioa701_snd_device); - if (!ret) - return 0; - - platform_device_put(mioa701_snd_device); - return ret; + mioa701.dev = &pdev->dev; + rc = snd_soc_register_card(&mioa701); + if (!rc) + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + " Do not use without a good knowledge of mio's board design!\n"); + return rc; } static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) { - platform_device_unregister(mioa701_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; } -- 1.7.10

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[alsa-devel] [PATCH] ASoC: soc-dapm: Use ASYNC_DOMAIN_EXCLUSIVE
by Fabio Estevam 01 Jul '12

01 Jul '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit e895407d (async: introduce 'async_domain' type) changed the parameter types of async_schedule_domain() and async_synchronize_full_domain(). Use ASYNC_DOMAIN_EXCLUSIVE and fix the following build warnings: sound/soc/soc-dapm.c:1646:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1647:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type sound/soc/soc-dapm.c:1660:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1661:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Build tested only. sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9670668..c93e832 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1572,7 +1572,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) struct snd_soc_dapm_context *d; LIST_HEAD(up_list); LIST_HEAD(down_list); - LIST_HEAD(async_domain); + ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; trace_snd_soc_dapm_start(card); -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 29 Jun '12

29 Jun '12

On 06/29/2012 05:34 AM, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. > > Because generic soc-dmaengine-pcm uses the DMAs API > based on dmaengine, using the exported APIs provided > by this generic driver. > > The new driver is selected if legacy driver is not > selected and new dma driver is enabled through config > file. This works just fine with the existing non-dmaengine DMA driver enabled. However, I can't get it to work with dmaengine: > # aplay ~/abba-dq-48000-stereo.wav > [ 151.613476] tegra20-i2s tegra20-i2s.0: dmaengine pcm open failed with err -6 > [ 151.620557] tegra20-i2s tegra20-i2s.0: can't open platform tegra20-i2s.0: -6 > aplay: main:654: audio open error: No such device or address I do have the following in my local tree: 68a67b8 ARM: tegra: add device tree AUXDATA for APBDMA 0db7a96 ARM: tegra: dma: rename driver name for clock to "tegra-apbdma" I also fixed the compatible values in drivers/dma/tegra20-apb-dma.c so the driver would get instantiated, which it does; /sys/devices/tegra-apbdma/dma has a bunch of dmaengine channels in it. (Note: This is on Ventana, although I doubt that makes much difference) Is there something else I need to do to test this?

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 29 Jun '12

29 Jun '12

At Thu, 28 Jun 2012 19:06:07 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:58 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:34:02 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > >>> At Thu, 28 Jun 2012 18:05:53 +0530, > >>> Laxman Dewangan wrote: > >>>> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>>>> * PGP Signed by an unknown key > >>>>> > >>>>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>>>> there any effort being made to make this generally available? > >>>>>> Yes there is, and in fact in next/master there is generic > >>>>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>>>> did a lot of work on this, I've put him on Cc. > >>>> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >>>> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >>>> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >>>> directly use that one? > >>>> Or, wait for this common API until all ARCH support it? > >>> I think it's fine to put it first in ALSA side with some ifdef. > >>> A similar trick is already found for snd_pcm_lib_mmap_iomem. > >>> See include/sound/pcm.h. > >>> > >>> But, actually it's a still question what if an architecture doesn't > >>> support the mmap of writecombine at all. The proposed patch doesn't > >>> allow you to know whether writecombine-mmap is possible or not on the > >>> running architecture until you really try to call it and fail. > >>> It's a missing piece, IMO. > >> Lars wanted to move the buffer allocation also to common place. > > Which common place? > > He wanted to refactor allocation part also at same time and so I guessed > that it will be in some common place and though about the sound/core. But the memory allocation rework is basically irrelevant from what's needed for pcm_mmap_writecombine() now. It'd need a lot more reworks than the simple addition of dma_mmap_writecombine() wrapper. > >> Then how about this? > >> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >> this in sound/core. > >> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >> which will be select this config through sound/soc/xxx/Kconfig if they > >> want to use. > >> This will provide three apis: new, free and mmap. > > Way too much hustles than necessary... > Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > Bit I like to hear simple way which solves the purpose. As Lars suggested, a simple ifdef should suffice for now. With the upcoming generic dma_mmap_writecombine() stuff, it'd be even simpler in future. But, still we need to be careful about this. As mentioned, there is no flag to know the possibility of writecombine mmap beforehand. It'd be nice if we have either a compile-time or a run-time flag / function to check that. Then the driver can also expose the mmap capability to user-space depending on the flag. Takashi

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Lars-Peter Clausen 29 Jun '12

29 Jun '12

On 06/29/2012 05:04 PM, Laxman Dewangan wrote: > [...] > > >> +/* allocate the coherent DMA pages */ >> +static void *snd_malloc_dev_pages(struct device *dev, size_t size, >> dma_addr_t *dma) >> +{ >> + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) > > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } I think all dma_alloc_* functions use dma_alloc_attrs internally these days. I think it might be a better idea just to use dma_alloc_attrs in __snd_malloc_dev_pages and just pass a dma_attrs struct instead of a function. Something like: static void *__snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma, struct dma_attrs *attrs) { .... res = dma_alloc_attrs(dev, PAGE_SIZE << pg, dma, gfp_flags, attrs); .... } static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { return __snd_malloc_dev_pages(dev, size, dma, NULL); } static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) { DEFINE_DMA_ATTRS(attrs); dma_set_attr(DMA_ATTR_WRITE_COMBINE, &attrs); return __snd_malloc_dev_pages(dev, size, dma, &attrs); } Same applies for freeing the memory again. - Lars

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 20:34:24 +0530, Laxman Dewangan wrote: > > Hi Takashi, > Thanks for sample code. It helps lot. > > On Friday 29 June 2012 05:43 PM, Takashi Iwai wrote: > > At Fri, 29 Jun 2012 15:53:15 +0530, > > Laxman Dewangan wrote: > >> Some of the ARM based soc allocate the writecombine dma buffer for > >> pcm substreams. They have the same codes for managing this buffer. > >> Moving this to the core/pcm files so that they can use that directly. > >> > >> Remove the code from Tegra PCM and use these new library function. > >> > >> This is enabled only for ARM specific and can be extended to other > >> architecture if they support the writecombine dma buffer. > >> > >> This patch is based on detail discussion on patch: > >> [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > >> And suggestion from Lars and Takashi. > > Looking through your patch, I think an easier integration is just to > > add writecombine option to memalloc.c which calls > > dma_alloc_writecombine() instead of dma_alloc_coherent(). > > > > The addition for mmap is still an open question. Again, an easier > > option so far looks like just add the call of dma_alloc_writecombine() > > in snd_pcm_lib_default_mmap(). Alternatively, we can create an > > individual mmap pcm_ops as we discussed. > > > > Below is a totally untested patch, but you can imagine what I meant. > > > > After this change, replace with snd_pcm_lib_malloc_pages with > > SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. > > We can not use the snd_pcm_lib_malloc_pages() in the pcm_new callback > because at this time the substream->runtime is not initialized and it > leads to the kernel crash. > > I used the apis as > > int snd_soc_pcm_new_wc_dma_buffer(struct snd_soc_pcm_runtime *rtd, > size_t max_bytes) > { > ::::::::::::: > substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; > if (substream) { > substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_WC; > ret = snd_pcm_lib_malloc_pages(substream, max_bytes); > if (ret) > goto err; > } > :::::::: > } > > > int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t > size) > { > struct snd_pcm_runtime *runtime; > struct snd_dma_buffer *dmab = NULL; > > if (PCM_RUNTIME_CHECK(substream)) > return -EINVAL; > > if (snd_BUG_ON(substream->dma_buffer.dev.type == > SNDRV_DMA_TYPE_UNKNOWN)) > return -EINVAL; > > runtime = substream->runtime; > > if (runtime->dma_buffer_p) { > ---------------Kernel crash at this point ---------- > > So I used the snd_dma_alloc_pages() from the driver to allocate WC memory. snd_pcm_lib_malloc_pages() should be called only from hw_params. Prior to that, you need to set up the dam_buffer type by calling snd_pcm_lib_preallocate_pages() or snd_pcm_lib_preallocate_pages_for_all() at the time to create a PCM instance. The size can be 0 if not necessary to allocate there but later on demand. > > > +/* allocate the coherent DMA pages */ > > +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) > > +{ > > + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) Ah, OK. thanks, Takashi > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } >

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 15:53:15 +0530, Laxman Dewangan wrote: > > Some of the ARM based soc allocate the writecombine dma buffer for > pcm substreams. They have the same codes for managing this buffer. > Moving this to the core/pcm files so that they can use that directly. > > Remove the code from Tegra PCM and use these new library function. > > This is enabled only for ARM specific and can be extended to other > architecture if they support the writecombine dma buffer. > > This patch is based on detail discussion on patch: > [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > And suggestion from Lars and Takashi. Looking through your patch, I think an easier integration is just to add writecombine option to memalloc.c which calls dma_alloc_writecombine() instead of dma_alloc_coherent(). The addition for mmap is still an open question. Again, an easier option so far looks like just add the call of dma_alloc_writecombine() in snd_pcm_lib_default_mmap(). Alternatively, we can create an individual mmap pcm_ops as we discussed. Below is a totally untested patch, but you can imagine what I meant. After this change, replace with snd_pcm_lib_malloc_pages with SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. Takashi --- diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c425062..7840340 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_WC 4 /* writecombine pages */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6915692..11108e8 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -124,8 +124,12 @@ void snd_free_pages(void *ptr, size_t size) */ #ifdef CONFIG_HAS_DMA -/* allocate the coherent DMA pages */ -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +static void *__snd_malloc_dev_pages(struct device *dev, size_t size, + dma_addr_t *dma, + void *(*func)(struct device *dev, + size_t size, + dma_addr_t *dma_handle, + gfp_t flag)) { int pg; void *res; @@ -138,16 +142,18 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d | __GFP_COMP /* compound page lets parts be mapped */ | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); + res = func(dev, PAGE_SIZE << pg, dma, gfp_flags); if (res != NULL) inc_snd_pages(pg); return res; } -/* free the coherent DMA pages */ -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, - dma_addr_t dma) +static void __snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma, + void (*func)(struct device *dev, + size_t size, void *vaddr, + dma_addr_t dma_handle)) { int pg; @@ -155,8 +161,36 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, return; pg = get_order(size); dec_snd_pages(pg); - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); + func(dev, PAGE_SIZE << pg, ptr, dma); +} + +/* allocate the coherent DMA pages */ +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); +} + +/* free the coherent DMA pages */ +static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_coherent); } + +#ifdef CONFIG_ARM +/* allocate the writecombine DMA pages */ +static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_writecombine); +} + +/* free the writecombine DMA pages */ +static void snd_free_dev_wc_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_writecombine); +} +#endif /* CONFIG_ARM */ #endif /* CONFIG_HAS_DMA */ /* @@ -200,6 +234,11 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + dmab->area = snd_malloc_dev_wc_pages(device, size, &dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -272,6 +311,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + snd_free_dev_wc_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -378,7 +422,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "WC" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c499791 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3170,6 +3170,14 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_RESERVED; +#ifdef CONFIG_ARM + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_WC) + return dma_mmap_writecombine(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV)

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC settings
by Prchal Jiří 29 Jun '12

29 Jun '12

This patch adds AGC target level and times settings for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 50a7112..5335ebc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -204,6 +204,10 @@ static const char *aic3x_right_hpcom_mux[] = static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -231,6 +235,21 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +#define LAGC_LEV_ENUM 0 +#define RAGC_LEV_ENUM 1 +#define LAGC_ATT_ENUM 2 +#define RAGC_ATT_ENUM 3 +#define LAGC_DEC_ENUM 4 +#define RAGC_DEC_ENUM 5 +static const struct soc_enum aic3x_agc_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -355,6 +374,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_agc_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, -- 1.7.7

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[alsa-devel] ASoC updates for 3.5
by Mark Brown 29 Jun '12

29 Jun '12

The following changes since commit 3419ae781f1592b3d367107db6500090495490cd: ASoC: tegra+wm8903: turn of mic detect when card is removed (2012-06-12 11:38:48 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 5e531fc87875fc0bbfb3ca46a8a178674c4e9b7c: ASoC: tlv320aic3x: Fix codec pll configure bug (2012-06-27 12:46:25 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.5 A couple of unexciting driver specific changes plus a fix for gain controls used as DAPM controls which isn't yet needed in mainline but is a definite bug. ---------------------------------------------------------------- Benoît Thébaudeau (1): ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect Hebbar, Gururaja (1): ASoC: tlv320aic3x: Fix codec pll configure bug Mark Brown (1): ASoC: wm2200: Add missing BCLK rate sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/wm2200.c | 1 + sound/soc/soc-dapm.c | 8 +------- 4 files changed, 4 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH] ASoC: spdif: Build separate RX and TX objects
by Mark Brown 29 Jun '12

29 Jun '12

Otherwise we fail to link when building as modules due to multiple init/exit functions. Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 62c3d4d..acf8088 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,7 +45,8 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o @@ -159,7 +160,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o -- 1.7.10

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Re: [alsa-devel] [pulseaudio-discuss] [PATCH 2/4] alsa-mixer: Add Phantom Jack support
by Raymond Yau 29 Jun '12

29 Jun '12

2012/6/28 David Henningsson <david.henningsson(a)canonical.com>: > For kernel 3.6, "phantom jack" kctls have been added. They serve as > a marker that a particular port exist. They were made so we can detect > that there actually are speakers and internal mic on a laptop, even if > there are no other indications (volume controls etc). > > Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> > --- It seem that there will be numerous phantom ports created e.g. two internal mic phantom ports, spdif and spdif in, cd on the intel d975 desktop ./hda-emu codecs/stac9274d-intel-d975 CTRL: add: Front Headphone Jack:0 JACK created Front Headphone, type 1 JACK report Front Headphone, status 0 CTRL: add: Front Mic Jack:0 JACK created Front Mic, type 2 JACK report Front Mic, status 0 CTRL: add: Rear Mic Jack:0 JACK created Rear Mic, type 2 JACK report Rear Mic, status 0 CTRL: add: Internal Mic Phantom Jack:0 CTL Notify: Internal Mic Phantom Jack:0, mask=1 CTRL: add: Internal Mic Phantom Jack:1 CTL Notify: Internal Mic Phantom Jack:1, mask=1 CTRL: add: Line Jack:0 JACK created Line, type 32 JACK report Line, status 0 CTRL: add: CD Phantom Jack:0 CTL Notify: CD Phantom Jack:0, mask=1 CTRL: add: SPDIF Phantom Jack:0 CTL Notify: SPDIF Phantom Jack:0, mask=1 CTRL: add: SPDIF In Phantom Jack:0 CTL Notify: SPDIF In Phantom Jack:0, mask=1

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[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 29 Jun '12

29 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ 3 files changed, 38 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..542d9a8 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..f0d23d6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..49d7b99 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio wallclock timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 19:30:57 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 07:21 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 19:06:07 +0530, > > Laxman Dewangan wrote: > > > >>>> Then how about this? > >>>> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >>>> this in sound/core. > >>>> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >>>> which will be select this config through sound/soc/xxx/Kconfig if they > >>>> want to use. > >>>> This will provide three apis: new, free and mmap. > >>> Way too much hustles than necessary... > >> Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > >> Bit I like to hear simple way which solves the purpose. > > As Lars suggested, a simple ifdef should suffice for now. > > With the upcoming generic dma_mmap_writecombine() stuff, it'd be even > > simpler in future. > > > > But, still we need to be careful about this. As mentioned, there is > > no flag to know the possibility of writecombine mmap beforehand. > > It'd be nice if we have either a compile-time or a run-time flag / > > function to check that. Then the driver can also expose the mmap > > capability to user-space depending on the flag. > > Yaah, this seems really simple. Thanks for suggestion. > If I understand it fully, the new apis will be declare in sound/pcm.h, > implement it in pcm_native.c and use the config variable > CONFIG_SND_PCM_WRITECOMBINE_BUFFER for ifdef. Or this could be simply ifdef CONFIG_ARM. Since this ifdef will be unnecessary soon later, I don't think we need to give yet another kconfig. Of course, we should put a good comment around there for not keeping it for long time. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 01:56 PM, Laxman Dewangan wrote: > On Thursday 28 June 2012 04:21 PM, Mark Brown wrote: >> * PGP Signed by an unknown key >> >> On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: >>> The pcm mmap for the substream dma buffer is done using the >>> dma_mmap_writecombine(). >>> Some of the chip sound driver like Spear, Tegra, Davinci, >>> ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this >>> mapping locally. >>> Add api in the snd dmaengine driver so that different client of >>> snd dmaengine can use this rather than implementing locally. >> OK, but I'd expect this to come along with one or more patches >> converting at least some of the users to the new API... > > This is effort towards moving the Tegra pcm driver for dmangine based dma > driver inplace of legacy dma driver for Tegra which have private API. > > Currently, following 4 pcm driver can use this API directly keeping that > they already using the generic snd dmaengine pcm driver. > ep93xx, > spear, > fsl, > mxs. > And Tegra will be next. > > Other driver which can use this API are davinci, OMAP, samsung but because > they are not using the snd_dmaengine_pcm driver's api as of now so not be > very much useful for them. Btw. all these drivers basically use the same code for buffer preallocation. Since how the buffer is mmaped is related to how it is allocated it might be worth to factor this out at the same time as well. - Lars

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:34:02 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:05:53 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>> * PGP Signed by an unknown key > >>> > >>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>> there any effort being made to make this generally available? > >>>> Yes there is, and in fact in next/master there is generic > >>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>> did a lot of work on this, I've put him on Cc. > >> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >> directly use that one? > >> Or, wait for this common API until all ARCH support it? > > I think it's fine to put it first in ALSA side with some ifdef. > > A similar trick is already found for snd_pcm_lib_mmap_iomem. > > See include/sound/pcm.h. > > > > But, actually it's a still question what if an architecture doesn't > > support the mmap of writecombine at all. The proposed patch doesn't > > allow you to know whether writecombine-mmap is possible or not on the > > running architecture until you really try to call it and fail. > > It's a missing piece, IMO. > > Lars wanted to move the buffer allocation also to common place. Which common place? > Then how about this? > create new file and header for snd-pcm-writecombine-buffer.c/.h and put > this in sound/core. > Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > which will be select this config through sound/soc/xxx/Kconfig if they > want to use. > This will provide three apis: new, free and mmap. Way too much hustles than necessary... Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:05:53 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > > * PGP Signed by an unknown key > > > > On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>> Oh, dear - this means we don't actually have a generic API at all. Is > >>> there any effort being made to make this generally available? > >> Yes there is, and in fact in next/master there is generic > >> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >> though. But in theory it should fallback to a normal coherent mapping. Marek > >> did a lot of work on this, I've put him on Cc. > > So can we put the function snd_pcm_lib_writecombine_mmap() in the > pcm_native.c and only export this api for ARM i.e. under macro #ifdef > CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > directly use that one? > Or, wait for this common API until all ARCH support it? I think it's fine to put it first in ALSA side with some ifdef. A similar trick is already found for snd_pcm_lib_mmap_iomem. See include/sound/pcm.h. But, actually it's a still question what if an architecture doesn't support the mmap of writecombine at all. The proposed patch doesn't allow you to know whether writecombine-mmap is possible or not on the running architecture until you really try to call it and fail. It's a missing piece, IMO. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 09:01 AM, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. > This is not really related to the dmaengine pcm driver. It's more of a coincidence that all upstream drivers which use the dmaengine pcm driver also use write-combined memory. In my opinion it would be better to add this to the ALSA core. Now that there is a generic dma_mmap_writecombine it may make sense to integrate this with snd_pcm_lib_default_mmap. - Lars > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > include/sound/dmaengine_pcm.h | 2 ++ > sound/soc/soc-dmaengine-pcm.c | 17 +++++++++++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h > index b877334..15c0d3c 100644 > --- a/include/sound/dmaengine_pcm.h > +++ b/include/sound/dmaengine_pcm.h > @@ -47,4 +47,6 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream); > > struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); > > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma); > #endif > diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c > index 5df529e..6535eb3 100644 > --- a/sound/soc/soc-dmaengine-pcm.c > +++ b/sound/soc/soc-dmaengine-pcm.c > @@ -21,6 +21,7 @@ > #include <linux/module.h> > #include <linux/init.h> > #include <linux/dmaengine.h> > +#include <linux/dma-mapping.h> > #include <linux/slab.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -313,3 +314,19 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) > return 0; > } > EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); > + > +/** > + * snd_dmaengine_pcm_mmap_writecombine - mmap stream dma buffer using > + * dma_mmap_writecombine(). > + * @substream: PCM substream > + * @vma: VM area. > + */ > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + > + return dma_mmap_writecombine(substream->pcm->card->dev, vma, > + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); > +} > +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_mmap_writecombine);

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